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From: Rashad Tatum <tatum.rashad@gmail.com>
To: ffmpeg-devel@ffmpeg.org
Date: Tue, 3 Jun 2025 12:23:28 -0400
Message-ID: <20250603162332.3230398-1-tatum.rashad@gmail.com>
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Subject: [FFmpeg-devel] [PATCH v3] Add new method for playing network based
streams with a play rate.
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Add implementation for changing the play rate for rtsp streams.
---
libavformat/avformat.h | 6 ++++++
libavformat/demux.h | 6 ++++++
libavformat/demux_utils.c | 6 ++++++
libavformat/rtsp.c | 1 +
libavformat/rtsp.h | 10 ++++++++++
libavformat/rtspdec.c | 21 +++++++++++++++++++--
6 files changed, 48 insertions(+), 2 deletions(-)
diff --git a/libavformat/avformat.h b/libavformat/avformat.h
index 2034d2aecc..7819b61d3e 100644
--- a/libavformat/avformat.h
+++ b/libavformat/avformat.h
@@ -2358,6 +2358,12 @@ int avformat_flush(AVFormatContext *s);
*/
int av_read_play(AVFormatContext *s);
+/**
+ * Play a network-based stream (e.g. RTSP stream) with a given play rate
+ * (e.g. Scale value for RTSP) and timestamp position.
+ */
+int av_read_play_with_rate(AVFormatContext *s, double play_rate, int stream_index, int64_t timestamp);
+
/**
* Pause a network-based stream (e.g. RTSP stream).
*
diff --git a/libavformat/demux.h b/libavformat/demux.h
index e83d84a201..965743bdf0 100644
--- a/libavformat/demux.h
+++ b/libavformat/demux.h
@@ -113,6 +113,12 @@ typedef struct FFInputFormat {
* (RTSP).
*/
int (*read_play)(struct AVFormatContext *);
+
+ /**
+ * Play a network-based stream (e.g. RTSP stream) with a given play rate
+ * (e.g. Scale value for RTSP) and timestamp position.
+ */
+ int (*read_play_with_rate)(struct AVFormatContext *s, double play_rate, int stream_index, int64_t timestamp);
/**
* Pause playing - only meaningful if using a network-based format
diff --git a/libavformat/demux_utils.c b/libavformat/demux_utils.c
index b632277460..8fed41e3b6 100644
--- a/libavformat/demux_utils.c
+++ b/libavformat/demux_utils.c
@@ -179,6 +179,12 @@ int av_read_play(AVFormatContext *s)
return AVERROR(ENOSYS);
}
+int av_read_play_with_rate(AVFormatContext *s, double play_rate, int stream_index, int64_t timestamp) {
+ if (ffifmt(s->iformat)->read_play_with_rate)
+ return ffifmt(s->iformat)->read_play_with_rate(s, play_rate, stream_index, timestamp);
+ return AVERROR(ENOSYS);
+}
+
int av_read_pause(AVFormatContext *s)
{
if (ffifmt(s->iformat)->read_pause)
diff --git a/libavformat/rtsp.c b/libavformat/rtsp.c
index 5ea471b40c..6cfbcdf989 100644
--- a/libavformat/rtsp.c
+++ b/libavformat/rtsp.c
@@ -1998,6 +1998,7 @@ redirect:
av_strlcpy(rt->real_challenge, real_challenge, sizeof(rt->real_challenge));
rt->state = RTSP_STATE_IDLE;
rt->seek_timestamp = 0; /* default is to start stream at position zero */
+ rt->scale = 1.0; /* default is to play at the normal rate */
return 0;
fail:
ff_rtsp_close_streams(s);
diff --git a/libavformat/rtsp.h b/libavformat/rtsp.h
index 83b2e3f4fb..1b8ce05d74 100644
--- a/libavformat/rtsp.h
+++ b/libavformat/rtsp.h
@@ -245,6 +245,16 @@ typedef struct RTSPState {
* whenever we resume playback. Either way, the value is only used once,
* see rtsp_read_play() and rtsp_read_seek(). */
int64_t seek_timestamp;
+
+ /** the scale value requested when calling av_read_play(). This value
+ * is subsequently used as part of the "Scale" parameter when emitting
+ * the RTSP PLAY command. The "Scale" parameter determines the stream play
+ * rate. A value of 1 represents the normal play rate. Any other value is
+ * in regards to the normal play rate. A negative value represents reverse
+ * playback. If we are currently playing, this command is called instantly.
+ * If we are currently paused, this command is called whenever we resume
+ * playback. */
+ double scale;
int seq; /**< RTSP command sequence number */
diff --git a/libavformat/rtspdec.c b/libavformat/rtspdec.c
index 10078ce2fa..f1e44ccb63 100644
--- a/libavformat/rtspdec.c
+++ b/libavformat/rtspdec.c
@@ -39,6 +39,7 @@
#include "tls.h"
#include "url.h"
#include "version.h"
+#include <stdio.h>
static const struct RTSPStatusMessage {
enum RTSPStatusCode code;
@@ -527,7 +528,7 @@ static int rtsp_read_play(AVFormatContext *s)
{
RTSPState *rt = s->priv_data;
RTSPMessageHeader reply1, *reply = &reply1;
- int i;
+ int i, cmd_char_count = 0;
char cmd[MAX_URL_SIZE];
av_log(s, AV_LOG_DEBUG, "hello state=%d\n", rt->state);
@@ -564,11 +565,17 @@ static int rtsp_read_play(AVFormatContext *s)
if (rt->state == RTSP_STATE_PAUSED) {
cmd[0] = 0;
} else {
- snprintf(cmd, sizeof(cmd),
+
+ cmd_char_count += snprintf(cmd, sizeof(cmd),
"Range: npt=%"PRId64".%03"PRId64"-\r\n",
rt->seek_timestamp / AV_TIME_BASE,
rt->seek_timestamp / (AV_TIME_BASE / 1000) % 1000);
+
+ snprintf(cmd + cmd_char_count, sizeof(cmd) - cmd_char_count, "Scale: %f\r\n", rt->scale);
+
}
+
+
ff_rtsp_send_cmd(s, "PLAY", rt->control_uri, cmd, reply, NULL);
if (reply->status_code != RTSP_STATUS_OK) {
return ff_rtsp_averror(reply->status_code, -1);
@@ -593,6 +600,15 @@ static int rtsp_read_play(AVFormatContext *s)
return 0;
}
+static int rtsp_read_play_with_rate(AVFormatContext *s, double play_rate, int stream_index, int64_t timestamp) {
+ RTSPState *rt = s->priv_data;
+ rt->seek_timestamp = av_rescale_q(timestamp,
+ s->streams[stream_index]->time_base,
+ AV_TIME_BASE_Q);
+ rt->scale = play_rate;
+ return rtsp_read_play(s);
+}
+
/* pause the stream */
static int rtsp_read_pause(AVFormatContext *s)
{
@@ -1006,5 +1022,6 @@ const FFInputFormat ff_rtsp_demuxer = {
.read_close = rtsp_read_close,
.read_seek = rtsp_read_seek,
.read_play = rtsp_read_play,
+ .read_play_with_rate = rtsp_read_play_with_rate,
.read_pause = rtsp_read_pause,
};
--
2.49.0
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